The invention relates to digital coding of audio signals, in particular for producing 32 Kb/s data useful in telecommunications applications.
2. Background Information
In telecommunications, signals are nowadays mostly transmitted in digital form.
To meet the increasing need for connections with existing line capacities necessitates decreasing bit rates with unchanged intelligibility.
For good intelligibility, e.g., during hands-free operation or in video conferences, the present bandwidth of 300 to 3400 Hz at 64 kbs is already insufficient. The goal is to achieve good speech quality (also for music signals) for future bandwidths of 50 to 7000 Hz with a simultaneous reduction of the data rate to 32 kbs.
A coding technique for wideband (0 to 20 kHz) signals is known (R. Orglmeister, "Transformationscodierung mit fester Bitzuteilung bei Audiosignalen", FREQUENZ 44 (1990) 9-10, pp. 226-232) in which a sequence of sample values (44.1-kHz sampling with 16 bits per sample value) is first divided into blocks of equal size and processed by overtapping windowing (rectangular cosine function) to suppress audible block-boundary effects.
The signal is then transformed into the frequency domain by means of a first Fourier transform and decomposed into magnitude and phase values.
The phase values are uniformly quantized, while the magnitude values are subjected to data reduction.
For the data reduction, magnitude groups are formed such that, instead of the individual magnitude values, only the geometric mean of all magnitudes of a group is transmitted.
These magnitude groups are then routed, according to frequency range, to five different quantizers which quantize nonuniformly, approximating the logarithmic loudness perception.
By this known method, good speech quality is achieved, but due to insufficient data reduction of the magnitude values, a data rate of 32 kbs is not attainable.